Posts: 18

Topic: Intercall VoIP/bridge Issues

I have some questions regarding adobe audio integration with Bridges and VoIP. Currently we are using an Intercall a res plus bridge for meetings.

However one major problem we have is with start the conference if the host is using VoIP. Does the host always have to call into bridge start the meeting? Is there a possible work around for this or will we have to have someone call in to the bridge in order to start the audio. Is there anyway to start audio without having to do this.

Re: Intercall VoIP/bridge Issues

I'm pretty certain the host has to call into the Intercall system to make the necessary connection and in a sense authorize the session. It's that way with MeetingOne and PGi as well. If you have an idea please submit it here: https://www.adobe.com/cfusion/mmform/in … e=wishform

Re: Intercall VoIP/bridge Issues

One way to solve this, is to have the Connect room use the Host pin to call into the bridge instead of the participant pin. That way the room is always your phone bridge host. However this can be an issue if your phone bridge doesn't allow more than one host to dial in at any give time.

Re: Intercall VoIP/bridge Issues

Is this limitation an attempt by these companies to make more money from the customer by forcing us to use telephone audio or is it simply a lack of technology on their part?

We recently moved our audio contract to PGi and experienced this deficiency. Our previous reseller (a much smaller operation) allowed for hosts to connect to the audio bridge and start the meeting using VOIP.

It is strange that no action has been taken to fix the issue because it is (for us) the one major advantage of WebEx over connect.

Re: Intercall VoIP/bridge Issues

The limitation comes from a phone bridge needing a host to be present to start it. This is, and has been, a primary function of phone bridges. I'm not sure where you are seeing a lack in technology.

The functionality within Connect should be the same irregardless of who you purchase it from. You can also have Adobe Support activate the VoIP integration with a fully integrated PGi, MeetingOne or Intercall bridge, meaning you don't have to tie it in as a Universal Voice bridge. That should solve the problem as well.

Re: Intercall VoIP/bridge Issues

I spoke to Adobe and they have explained it is a PGI issue which makes sense.

From your comments I believe it is therefore the PGi settings that are preventing us from having the host start a meeting from within Adobe using VOIP.

This is because the options "url", "dial in numbers" or "dial in steps" seen through the link below are not available to us in our PGi set up i.e. they must be hidden.

Further evidence is provided when I set up a new audio provider in my adobe back end and use the PGi details (this allows me the options above). Instead of inserting the participant code in the DTMF field I use the host code and it works perfectly.

Can you think of any reason why PGi would not allows this as standard or after severe pressing from us?

Re: Intercall VoIP/bridge Issues

Sadly that is a question for PGi, and they have some reps that do chime in here from time to time. I find that the other providers have a, shall we say, cleaner integration experience.

Did you try activating the VoIP integration with a fully integrated phone bridge? This way you don't set up the UV steps, but the room calls into the bridge with the intended integration and people can still have the choice of VoIP or phone?

Re: Intercall VoIP/bridge Issues

Yes, I believe this is what I did this afternoon (although could easily be wrong) by creating a new audio profile and inserting the host code in the appropriate DTMF step. By connecting the meeting to the audio bridge it dials the toll free number followed by the correct host code and connects immediately.

Re: Intercall VoIP/bridge Issues

Re: Intercall VoIP/bridge Issues

Jorma is correct. PGI has requested that Adobe hosting team not enable Universal Voice by default on new accounts because it increases your per minute costs. You have to call into Adobe Support and ask them to enable UV on your account. I tried to reverse this decision but PGI demanded this because they did not want customers getting irritated because their costs are higher with UV on. That will solve the problem. This is peculiar to PGI and not the other vendors.

Re: Intercall VoIP/bridge Issues

Hi Heywardjr,

I have already spoken to Adobe Support and they have confirmed that UV is turned on for our account and advised me to contact PGi.

It is therefore likely that PGi have created their Audio Provider Profile on our account to automatically enter the participant code of the user in the DTMF step of the "Dial-In steps", rather than the host. Adobe explained that this is a PGi setting which they must change.

As you say, this looks to be standard practice with PGi because when I try to explain what we are discussing here to their support, they doesn't seem to understand. I will continue escalating until resolved and post back here.

Re: Intercall VoIP/bridge Issues

Lets break this down:1-WebEx (saw that mentioned here...) charges for "PC Voice" (yeah, it's voip, but only in that your VOICE is on the internet... there is no quality of service, etc...) - between 0.03 and 0.06 per minute per connected user, and, depending on the version of WebEx, there are limits to how many can join your conference in this way.2 -WebEx only recently released the ability to Telephony partners to do "hybrid" PC to Telephone bridging. PGI, and I'm sure our competitors in the space, are working on how this best to be implemented.

Now that we're DONE with the webex talk... Another thing I saw in this thread - Your PRESENTER is the most important person on the call - so WHY would you just throw his/her voice packets out on the internet and hope people get them?If a listener on PC audio drops a packet, that's fine - the other 200 people probably did not. but if the PRESENTERS audio drops.. .that impacts all 200.... bad meeting experience. But that's just an aside...

Adobe uses the term "Universal Voice" for TWO DIFFERENT (but similar) things.

With Connect 7.5 they introduced the term - Universal Voice. Universal voice lets you create a "provider" that any user can use (admin level), or, a provider at the user level, that can then create/spawn profiles with specific Dialing Rules.

This "Universal Voice" allowed you to create an audio profile that would 1)Record from the bridge audio line, and 2) broadcast that Audio conference to users listening only on their Computer.

Along comes Connect 8.Adobe adds a feature that provides/supports full two way / Hybrid PC to Teleconference audio.Not sure who made the choice, but this too is called "Universal Voice". *sigh*

At PGI, we welcome this new feature - PC Bridged Audio (as we call it for purpose of differentiation...) because it allows our customers the ultimate flexibility.

You see, this new "feature" works with the fully integrated Telephone conferences that you've come to love - the Call Me Back feature, audio controls on the Attendee list, call out to audio only attendee, etc.

So, you can have a meeting and let folks choose to listen by PC, dial into conference, get called by the bridge, etc. You know who's who in the attendee list due to the audio presence features, can mute those pesky callers who put you on HOLD - pumping in Celene Dion music or worse..., and so on.

But Adobe, again their code not PGI, created a small problem (and we're working with our partners at Adobe to resolve this - trust me...)If this feature is enabled on you account, then EVERY CONFERENCE you have where you "Start Audio Conference" - gets called out to by the Adobe server.

What this means is, you and your friend George are doing a dry run. You start the audio conference and you get called back by the bridge. Adobe calls in to. So, instead of just 2 "audio calls" you're paying for 3. And this happens EVERY TIME you start the audio conference, when this feature is enabled.

What's the UPSIDE?1 - the recording is pulled directly onto the Adobe server, so there is no waiting for the audio files to merge as there was before2 - if you have 100 people on PC Audio, well, that's pretty cheap - pay for 1 dial in, get 100??3 - there is no limit on the number of PC audio attendees, up to the capacity of the meeting license type you're using. So you can have a Seminar Room Meeting, use your PGI "on demand" GlobalMeet audio account, have full integration and "call me back" capabilities, and have 1000 people on the conference - if you choose.... 300 on phones, the rest listening / participating on PC.

What's the Down?1 - That extra dial in. It exists. Even if you don't want to enable any of the advanced features (recording, broadcast, PC bridged audio, etc;) 2 - Sometimes Adobe does have issues with their "dialout" to our bridges. They've isolated this to one of their providers, and the issues have dropped precipitously in the last 4-5 months, but just like a telephone line can "drop", so can the adobe dial-in. (BUT - You can enable a backup recording with PGI - in the Administrative controls - when this "PC Bridged Audio" is enabled)

Since the debut of this feature we've been after Adobe product management to change the workflow of the dialout to the bridge. This dialout should ONLY happen if/when the HOST chooses "Broadcast Audio" or "Record"... This has been documented and submitted to Adobe for review for quite some time, and our understanding is that they are working on this for an upcoming release. When that happens, it will be a great thing for our customers.

In the interim, because PGI cares about our customers and never wants to be seen as "Padding The Bill", we don't turn this on by default.

Your sales rep should be able to provide you this same "commentary", and request that you agree to the extra dial-in. Because it's an Adobe work flow issue, we can't credit the cost of that extra ghost dialer back, but we can have the full "PC Bridged Audio" capability turned on for any customer who understands and agrees with how this all works.

@tamborine_man -please reach out to your PGI sales team. If you need a walk-through of all the different capabilities discussed here, we'll make that happen.

@hayward - you know how to reach us. Please do so whenever you have the need.

Re: Intercall VoIP/bridge Issues

We are in contact with our PGi sales team who have passed us to support reps who have now passed us to the engineers.

Both the sales team and support reps have insisted that a host telephone dial in is the only method for linking VOIP and telephones.

While I believe they are doing this out of ignorance, their insistence that the "PC Bridged Audio" does not exist has resulted in the perception that PGi are attempting to "pad the bill" by forcing a host caller on every call. In effect the exact opposite of what was aimed for by making it default to begin with.

I will continue to pursue this and update the thread with any progress.

Also, I have one more question. We like that PGi produces a webpage with our list of telephone numbers and the hosts participant code. However when you turn your meeting details into an Outlook 2007 email by pressing "Outlook Invitation" from within Adobe, the url which which opens in the Outlook email contains spaces which break the url making making the link "dead". We asked our support and they said use the outlook add on or copy and paste the url over. Is there any better (back end) solution to this? It seems strange that more customer would not have mentioned this to them previously.

Re: Intercall VoIP/bridge Issues

After a month of going back and forth with our account manager and tier 1 support I got an answer from the engineers as to why we cannot have VOIP hosts initiating our calls.

Apparently the feature is broken on the relevant EMEA bridge and is in the process of being fixed. We have therefore been placed on another bridge until this time (although this was never communicated to us). It appears we are the only people in EMEA who have requested this functionality so there has been no rush for a resolution.

When it is fixed we will be issued with new phone numbers and new host/participant codes which we can role out to our users (*sigh* - a second role out after a disastrous first won't go down well).